1. Field of the Invention
The present invention relates to computer system architectures and more particularly to audio and video telecommunications for collaboration over hybrid networks.
2. Description of the Related Art
Since their introduction in the early 1980's, audio/video conferencing systems (“video conferencing systems”) have enabled users to communicate between remote sites using telephone lines based on dedicated or switched networks. Recently, technology and products to achieve the same over Internet Protocol have been attempted. Many such systems have emerged on the marketplace. Such systems produce low-frame-rate and low quality communications due to the unpredictable nature of the Internet. Such connections have been known to produce long latencies with limited bandwidth, resulting in jerky video, dropped audio and loss of lip sync.
Therefore, most video conferencing solutions have relied on dedicated switched networks such as T1/T3, ISDN or ATM. These systems have the disadvantage of higher cost and complexity and a lack of flexibility due largely to interoperability issues and higher cost client equipment. High costs are typically related to expensive conferencing hardware and dedicated pay-per-minute communications usage. Most often these dedicated communications circuits are switched circuits which use a fixed bandwidth allocation.
In most prior art systems the public switched telephone network (PSTN) is used to transfer audio during conferencing and collaboration with remote parties. It is known that quality of audio reception is poor over typical prior art Internet protocol (IP) systems. Prior art audio/video conferencing systems which use IP networks for audio and video transport lack the ability to terminate audio to client end systems through both PSTN and IP networks. Thus, it is desirable to achieve a hybrid mix of audio and video data over PSTN and IP-based audio/video conferencing to achieve full duplex real-time operation for all conference participants.
Modem voice over IP telephony systems have used the H.323 standard from the international telecommunications union (1TU). The H.323 standard focuses on the transmission of audio and video information through the Internet or switched private networks. FIG. 1 illustrates a prior art H.323 system. The block diagram of FIG. 1 includes a number of major components, including the general Internet 435, Internet H.323 bridges or gateways 411, telecommunications PSTN 433 (Public Switched Telephone Network), wireless and land-line phone handsets 412/413, standard Internet router 453, an optional gatekeeper 205, a multipoint control unit 203, a standard local area network 457, a voice over IP server running the H.323 protocol 201, and multiple I/O and display terminals 455. FIG. 1 is an example of the prior art conferencing system used between hybrid networks connecting the PSTN and Internet. Hybrid networks are used to communicate audio on internal LAN and WAN networks as well as transfer of audio to the existing telephone or PSTN network. While the H.323 recommendation allows for video conferencing, the prior art systems use private switched networks to establish transport that require expensive H.323 bridges between dedicated networks and the PSTN. Each of the components in FIG. 1 serves this purpose to achieve audio telecommunications between multiple parties.
Referring again to FIG. 1, the components of FIG. 1 are interconnected as follows. Prior art technology uses PC or client terminals 455 connected through a local area network 457 to either a data server or a specialized audio/video server 201. The network server 201 contains the application necessary to generate the H.323 network protocol. The data server 201 may be connected to a local gatekeeper 205 that is responsible for management control functions. As known the gatekeeper 205 is responsible for various duties such as admission control, status determination, and bandwidth management. Data server 201 functions are specified and handled through the ITU-H.225.0RAS recommendations. In addition, management control unit (MCU) 203 is connected to the data server 201. The multipoint control unit of a 203 is required by the eight-step ITU-5 H.323 recommendation for flexibility to negotiate end points and determine compatible setups for any conference media correspondents. The multipoint control unit 203 enables communication between three or more end points. Similar to a multipoint bridge, the gatekeeper 205 and the multipoint control unit 203 are optional components of the H.323 enabled network. Another useful job of the multipoint control unit 203 is to determine whether to unicast or multicast the audio or video streams. As known by one skilled in the art, these decisions are dependent on the capability of the underlying network and the topology of the multipoint conference. The multipoint control unit 203 determines the capabilities of each client terminal 455 and status each of media stream.
Again referring to FIG. 1, a standard network router 453 is connected between the local area network 457 and the Internet 435. At the outer edges of the Internet, “points of presence” are located at multiple end points or call termination sites. Gateways 411 are used to the transcode the H.323 network information onto the PSTN 433. Standard telephone handsets 413 or wireless phones 412 are connected to the PSTN telephony system.
FIG. 2 illustrates the embodiment of the H.323 protocol stack 200, its components and their interfaces to the local area network computers at the network interface 300. The input and control devices 455 along with a local area network 457 of FIG. 1 are shown in FIG. 2, consisting of the audio input output block 452, the video input and output block 451, the system control unit and data collaboration unit 459. These input devices are largely responsible for the delivery of media data to the H.323 protocol stack 200 shown in FIG. 2.
Again referring to FIG. 2, the sub blocks of functionality that make up the H.323 protocol stack 200 is described. The H.323 protocol stack consists of an audio codec 214, and a video CoDec 213 connected to the audio/video input and output blocks 452 and 451, respectively. The audio and video CoDecs are responsible for compression and decompression of the audio and video sources. The real-time network protocol component 215 is connected to the audio video CoDecs and is also responsible for preparation of the media data for transport according to the RTP (real-time protocol) recommendations.
Again referring to the prior art system of FIG. 2, the H.323 protocol stack has a system control unit 459 which connects to multiple control blocks within the H.323 protocol stack 200. The system control unit connects to the RTC Protocol block 217 for real time transport of the control information used to set-up and tear down the conference. The system control unit 459 also connects to the call-signaling units 221 and 219 for call signaling protocols and media stream packetization application used for packet-based multimedia communications. The system control unit 459 also connects to the control signaling block 223 used for control of protocols for multimedia communications. Lastly, the H.323 recommendation defines a data collaboration capability as known and outlined in the T.120 data collaboration unit 225.
All of the defined blocks make up the H.323 protocol network interface to the Transport protocol and network interface unit 300 for transport of data through the modem or router 453 to the Internet 435.